Voice over Internet Protocol (VoIP) is one of rapidly advancing Internet Protocol (IP) telephony technologies. VoIP is used by IP phones capable of sending and receiving packets over an IP network. For example, an IP phone converts an analog voice signal of a user's speech into a digital signal, packetizes the obtained digital voice signal, and sends/receives the packets to/from an IP phone of another party over the IP network. Communication between such IP phones is controlled by a call control server located in a Local Area Network (LAN).
When the call control server receives a call from an IP phone through a LAN, it detects an IP phone on the receiving side based on communication parameters of an IP phone on the sending side, and notifies the IP phone on the receiving side that there is an incoming call. The communication parameters include a dynamically determined IP address and a port number. When the IP phone on the receiving side responds to the call, the call control server receives the response from the IP phone on the receiving side and establishes a call session between the IP phone on the receiving side and the IP phone on the sending side. Consequently, P2P (peer-to-peer) packet direct communication is started between the IP phone on the receiving side and the IP phone on the sending side.
In order to ensure certain or better communication quality (call quality), the IP phones occupy a communication bandwidth assigned in advance to send/receive packets. The IP phones send/receive various kinds of information necessary for call session control to/from the call session server and a call session gateway before starting a telephone conversation. Therefore, it takes a time until a call session is established, and, if a speaker starts talking before a call session is established, there is a possibility that the beginning of the speaker's voice is cut off.
Hence, there is an IP phone capable of preventing cut-off of the beginning of the speaker's voice immediately after the start of the conversation (see Japanese Laid-open Patent Publication No. 2007-19767). This IP phone stores the transmitted voice as Realtime Transport Protocol (RTP) packets in its buffer memory until a call session is established, and sends the packets in the stored order to an IP network. Thus, it is possible to prevent cut-off of the beginning of voice at the start of a conversation.